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[alsa-devel] Underruns in basic audio playback application
by Devin Heitmueller 04 Dec '09

04 Dec '09

Hello all, I am working on a very basic application that reads a capture device and outputs to a playback device. It's what you would traditionally accomplish with something like the following: arecord -D hw:1,0 -r 48000 -c 2 -f S16_LE | aplay - However, in this case, since the final result will be part of a TV watching application, I need to maintain less than 30ms of latency to preserve lipsync. The application I ended up with essentially creates two PCM streams (one for capture, one for playback), sets them up with the same parameters in terms of rate, format, channels, etc., and then has a loop of snd_pcm_readi() and snd_pcm_writei() calls. The application works, except I am getting a considerable amount of underruns on the playback device. Given my relative inexperience with alsa-lib, I suspect that I have misconfigured one of the parameters effecting buffering - buffer size, buffer time, period size, period time. Can anyone offer any constructive suggestions or tips on what these tuning parameters should be set to for a continuous 48000Khz, 2 channel stream of audio? Also, should the buffering configuration really be the same for both the capture and playback stream, or do I need to play some games such that one requires more/less buffering than the other? Thanks in advance, Devin -- Devin J. Heitmueller - Kernel Labs http://www.kernellabs.com

2 2

[alsa-devel] [PATCH] ASoC: tlv320dac33: Add support for regulator framework
by Ilkka Koskinen 04 Dec '09

04 Dec '09

Take the regulator framework in use for managing the power sources. Signed-off-by: Ilkka Koskinen <ilkka.koskinen(a)nokia.com> --- sound/soc/codecs/tlv320dac33.c | 92 ++++++++++++++++++++++++++++++++++------ 1 files changed, 79 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 9c8903d..5037454 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -30,6 +30,7 @@ #include <linux/platform_device.h> #include <linux/interrupt.h> #include <linux/gpio.h> +#include <linux/regulator/consumer.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -58,11 +59,19 @@ enum dac33_state { DAC33_FLUSH, }; +#define DAC33_NUM_SUPPLIES 3 +static const char *dac33_supply_names[DAC33_NUM_SUPPLIES] = { + "AVDD", + "DVDD", + "IOVDD", +}; + struct tlv320dac33_priv { struct mutex mutex; struct workqueue_struct *dac33_wq; struct work_struct work; struct snd_soc_codec codec; + struct regulator_bulk_data supplies[DAC33_NUM_SUPPLIES]; int power_gpio; int chip_power; int irq; @@ -297,28 +306,49 @@ static inline void dac33_soft_power(struct snd_soc_codec *codec, int power) dac33_write(codec, DAC33_PWR_CTRL, reg); } -static void dac33_hard_power(struct snd_soc_codec *codec, int power) +static int dac33_hard_power(struct snd_soc_codec *codec, int power) { struct tlv320dac33_priv *dac33 = codec->private_data; + int ret; mutex_lock(&dac33->mutex); if (power) { - if (dac33->power_gpio >= 0) { - gpio_set_value(dac33->power_gpio, 1); - dac33->chip_power = 1; - /* Restore registers */ - dac33_restore_regs(codec); + ret = regulator_bulk_enable(ARRAY_SIZE(dac33->supplies), + dac33->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", ret); + goto exit; } + + if (dac33->power_gpio >= 0) + gpio_set_value(dac33->power_gpio, 1); + + dac33->chip_power = 1; + + /* Restore registers */ + dac33_restore_regs(codec); + dac33_soft_power(codec, 1); } else { dac33_soft_power(codec, 0); - if (dac33->power_gpio >= 0) { + if (dac33->power_gpio >= 0) gpio_set_value(dac33->power_gpio, 0); - dac33->chip_power = 0; + + ret = regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), + dac33->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to disable supplies: %d\n", ret); + goto exit; } + + dac33->chip_power = 0; } - mutex_unlock(&dac33->mutex); +exit: + mutex_unlock(&dac33->mutex); + return ret; } static int dac33_get_nsample(struct snd_kcontrol *kcontrol, @@ -469,6 +499,8 @@ static int dac33_add_widgets(struct snd_soc_codec *codec) static int dac33_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + int ret; + switch (level) { case SND_SOC_BIAS_ON: dac33_soft_power(codec, 1); @@ -476,12 +508,19 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) - dac33_hard_power(codec, 1); + if (codec->bias_level == SND_SOC_BIAS_OFF) { + ret = dac33_hard_power(codec, 1); + if (ret != 0) + return ret; + } + dac33_soft_power(codec, 0); break; case SND_SOC_BIAS_OFF: - dac33_hard_power(codec, 0); + ret = dac33_hard_power(codec, 0); + if (ret != 0) + return ret; + break; } codec->bias_level = level; @@ -959,6 +998,9 @@ static int dac33_soc_probe(struct platform_device *pdev) /* power on device */ dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Bias level configuration has enabled regulator an extra time */ + regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies); + return 0; pcm_err: @@ -1039,7 +1081,7 @@ static int dac33_i2c_probe(struct i2c_client *client, struct tlv320dac33_platform_data *pdata; struct tlv320dac33_priv *dac33; struct snd_soc_codec *codec; - int ret = 0; + int ret, i; if (client->dev.platform_data == NULL) { dev_err(&client->dev, "Platform data not set\n"); @@ -1130,6 +1172,24 @@ static int dac33_i2c_probe(struct i2c_client *client, } } + for (i = 0; i < ARRAY_SIZE(dac33->supplies); i++) + dac33->supplies[i].supply = dac33_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(dac33->supplies), + dac33->supplies); + + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err_get; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(dac33->supplies), + dac33->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_enable; + } + ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); @@ -1149,6 +1209,10 @@ static int dac33_i2c_probe(struct i2c_client *client, return ret; error_codec: + regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies); +err_enable: + regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies); +err_get: if (dac33->irq >= 0) { free_irq(dac33->irq, &dac33->codec); destroy_workqueue(dac33->dac33_wq); @@ -1177,6 +1241,8 @@ static int dac33_i2c_remove(struct i2c_client *client) if (dac33->irq >= 0) free_irq(dac33->irq, &dac33->codec); + regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies); + destroy_workqueue(dac33->dac33_wq); snd_soc_unregister_dai(&dac33_dai); snd_soc_unregister_codec(&dac33->codec); -- 1.6.0.4

3 2

[alsa-devel] ALSA Control Questions
by Tobias Schneider 04 Dec '09

04 Dec '09

I've got some questions concerning the ALSA control interface... 1. What is a "simple" control? I would like to use the control interface to set more than just volume (e.g. I would like to set input gain, output attenuation, input channel selection, output channel selection, output channel activation...), so I am wondering if I should use SNDRV_CTL_ELEM_IFACE_CARD or SNDRV_CTL_ELEM_IFACE_MIXER, as they are obviously in context with "simple" control... 2. The more interesting question: as written in "Writing an alsa driver" callbacks are basically not atomic, but I am getting BUG: scheduling while atomic: amixer/0x00000001/430 Call Trace: .[..] in "put" callback?? So what is meant with "basically not atomic"?? What is allowed and what is not allowed, am I able to get hardware interrupts in this context?

3 7

[alsa-devel] [PATCH] ASoC: tlv320dac33: Add support for regulator framework
by Ilkka Koskinen 04 Dec '09

04 Dec '09

Take the regulator framework in use for managing the power sources. Signed-off-by: Ilkka Koskinen <ilkka.koskinen(a)nokia.com> --- sound/soc/codecs/tlv320dac33.c | 92 ++++++++++++++++++++++++++++++++++------ 1 files changed, 79 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 3ca8934..c2510bb 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -30,6 +30,7 @@ #include <linux/platform_device.h> #include <linux/interrupt.h> #include <linux/gpio.h> +#include <linux/regulator/consumer.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -58,11 +59,19 @@ enum dac33_state { DAC33_FLUSH, }; +#define DAC33_NUM_SUPPLIES 3 +static const char *dac33_supply_names[DAC33_NUM_SUPPLIES] = { + "AVDD", + "DVDD", + "IOVDD", +}; + struct tlv320dac33_priv { struct mutex mutex; struct workqueue_struct *dac33_wq; struct work_struct work; struct snd_soc_codec codec; + struct regulator_bulk_data supplies[DAC33_NUM_SUPPLIES]; int power_gpio; int chip_power; int irq; @@ -297,28 +306,49 @@ static inline void dac33_soft_power(struct snd_soc_codec *codec, int power) dac33_write(codec, DAC33_PWR_CTRL, reg); } -static void dac33_hard_power(struct snd_soc_codec *codec, int power) +static int dac33_hard_power(struct snd_soc_codec *codec, int power) { struct tlv320dac33_priv *dac33 = codec->private_data; + int ret; mutex_lock(&dac33->mutex); if (power) { - if (dac33->power_gpio >= 0) { - gpio_set_value(dac33->power_gpio, 1); - dac33->chip_power = 1; - /* Restore registers */ - dac33_restore_regs(codec); + ret = regulator_bulk_enable(ARRAY_SIZE(dac33->supplies), + dac33->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", ret); + goto exit; } + + if (dac33->power_gpio >= 0) + gpio_set_value(dac33->power_gpio, 1); + + dac33->chip_power = 1; + + /* Restore registers */ + dac33_restore_regs(codec); + dac33_soft_power(codec, 1); } else { dac33_soft_power(codec, 0); - if (dac33->power_gpio >= 0) { + if (dac33->power_gpio >= 0) gpio_set_value(dac33->power_gpio, 0); - dac33->chip_power = 0; + + ret = regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), + dac33->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to disable supplies: %d\n", ret); + goto exit; } + + dac33->chip_power = 0; } - mutex_unlock(&dac33->mutex); +exit: + mutex_unlock(&dac33->mutex); + return ret; } static int dac33_get_nsample(struct snd_kcontrol *kcontrol, @@ -470,6 +500,8 @@ static int dac33_add_widgets(struct snd_soc_codec *codec) static int dac33_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + int ret; + switch (level) { case SND_SOC_BIAS_ON: dac33_soft_power(codec, 1); @@ -477,12 +509,19 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) - dac33_hard_power(codec, 1); + if (codec->bias_level == SND_SOC_BIAS_OFF) { + ret = dac33_hard_power(codec, 1); + if (ret != 0) + return ret; + } + dac33_soft_power(codec, 0); break; case SND_SOC_BIAS_OFF: - dac33_hard_power(codec, 0); + ret = dac33_hard_power(codec, 0); + if (ret != 0) + return ret; + break; } codec->bias_level = level; @@ -960,6 +999,9 @@ static int dac33_soc_probe(struct platform_device *pdev) /* power on device */ dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Bias level configuration has enabled regulator an extra time */ + regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies); + ret = snd_soc_init_card(socdev); if (ret < 0) { dev_err(codec->dev, "failed to register card\n"); @@ -1048,7 +1090,7 @@ static int dac33_i2c_probe(struct i2c_client *client, struct tlv320dac33_platform_data *pdata; struct tlv320dac33_priv *dac33; struct snd_soc_codec *codec; - int ret = 0; + int ret, i; if (client->dev.platform_data == NULL) { dev_err(&client->dev, "Platform data not set\n"); @@ -1138,6 +1180,24 @@ static int dac33_i2c_probe(struct i2c_client *client, } } + for (i = 0; i < ARRAY_SIZE(dac33->supplies); i++) + dac33->supplies[i].supply = dac33_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(dac33->supplies), + dac33->supplies); + + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err_get; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(dac33->supplies), + dac33->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_enable; + } + ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); @@ -1157,6 +1217,10 @@ static int dac33_i2c_probe(struct i2c_client *client, return ret; error_codec: + regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies); +err_enable: + regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies); +err_get: if (dac33->irq >= 0) { free_irq(dac33->irq, &dac33->codec); destroy_workqueue(dac33->dac33_wq); @@ -1185,6 +1249,8 @@ static int dac33_i2c_remove(struct i2c_client *client) if (dac33->irq >= 0) free_irq(dac33->irq, &dac33->codec); + regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies); + destroy_workqueue(dac33->dac33_wq); snd_soc_unregister_dai(&dac33_dai); snd_soc_unregister_codec(&dac33->codec); -- 1.6.0.4

3 2

[alsa-devel] [PATCH v2] ASoC: DaVinci: Update suspend/resume support for McASP driver
by Chaithrika U S 04 Dec '09

04 Dec '09

Add clock enable and disable calls to resume and suspend respectively. Also add a member to the audio device data structure which tracks the clock status. Tested on DA850/OMAP-L138 EVM. For the purpose of testing, the patches[1] which add suspend-to-RAM support to DA850/OMAP-L138 SoC were applied. [1] http://linux.davincidsp.com/pipermail/davinci-linux-open-source/ 2009-November/016958.html Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies to ALSA GIT tree on branch topic/asoc at http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog; h=topic/asoc sound/soc/davinci/davinci-mcasp.c | 18 ++++++++++++++++-- sound/soc/davinci/davinci-mcasp.h | 1 + sound/soc/davinci/davinci-pcm.c | 2 +- 3 files changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 0a302e1..a613bbb 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -767,14 +767,27 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, int ret = 0; switch (cmd) { - case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + if (!dev->clk_active) { + clk_enable(dev->clk); + dev->clk_active = 1; + } + /* Fall through */ + case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: davinci_mcasp_start(dev, substream->stream); break; - case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: + davinci_mcasp_stop(dev, substream->stream); + if (dev->clk_active) { + clk_disable(dev->clk); + dev->clk_active = 0; + } + + break; + + case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: davinci_mcasp_stop(dev, substream->stream); break; @@ -866,6 +879,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) } clk_enable(dev->clk); + dev->clk_active = 1; dev->base = (void __iomem *)IO_ADDRESS(mem->start); dev->op_mode = pdata->op_mode; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 582c924..e755b51 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -44,6 +44,7 @@ struct davinci_audio_dev { int sample_rate; struct clk *clk; unsigned int codec_fmt; + u8 clk_active; /* McASP specific data */ int tdm_slots; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index ad4d7f4..80c7fdf 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -49,7 +49,7 @@ static void print_buf_info(int slot, char *name) static struct snd_pcm_hardware pcm_hardware_playback = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE), + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), .formats = (SNDRV_PCM_FMTBIT_S16_LE), .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | -- 1.5.6

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[alsa-devel] snd_hda_intel: MacBook Pro 5,1 / ALC889A issues
by Alexey Spiridonov 04 Dec '09

04 Dec '09

Hello, The hda_intel driver is only marginally usable on my laptop. I described the problems in detail here: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/488103 In brief, one has to choose between two sets of problems: 1) model=mbp3: broken speakers or 2) model=mb5: working speakers, but noisy headphone output Neither option provides headphone jack sense. Neither option is able to access the headset-type microphone available under MacOS through 3-band jacks like this: http://img.tomshardware.com/us/2007/07/19/how_good_is_apples_wired_iphone_h… (MacOS automatically switches from the in-computer mic to the the headset mic -- I haven't been able to get any signal from it despite trying all permutations of alsamixer's capture settings) I have attached the output of alsa-info.sh. I'd be happy to help test patches, or other debugging support. Thanks a lot! ~a

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[alsa-devel] syslog says snd-usb-audio bug?
by Achilles H 04 Dec '09

04 Dec '09

Dec 4 00:16:21 susbwoy pulseaudio[1719]: alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Dec 4 00:16:21 susbwoy pulseaudio[1719]: alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Dec 4 00:16:21 susbwoy pulseaudio[1719]: alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. I am using Ubuntu 9.10 64bit with an Edirol UA-25 USB sound card. I installed OpenArena, and tried to open it. Screen went black - Could only get back to the desktop by killing X. This is what happened in the terminal when OpenArena Died at Initializing sound ------ Initializing Sound ------ bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) Allocated 96 sources. OpenAL default capture device is 'ALSA Capture on default' AL lib: alsa.c:675: set access failed: Invalid argument AL lib: oss.c:307: Could not open /dev/dsp: No such file or directory Killed I have no issues with sound at all apart from trying to use OpenAL.Warsow(another great game) will also refuse to run with OpenAL, except it runs great in the default sound mode 'qf'. What is going on here? thanks

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[alsa-devel] [PATCH] ASoC: Rename controls with a / in wm_hubs
by Joonyoung Shim 04 Dec '09

04 Dec '09

This renames from a character / to : of controls. A / occurs below error messages. ASoC: Failed to create IN2RP/VXRP debugfs file ASoC: Failed to create IN2LP/VXRN debugfs file Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- If there is the better a delimiter or other idea, please inform me it. sound/soc/codecs/wm_hubs.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 810a563..d73c305 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -438,11 +438,11 @@ static const struct snd_soc_dapm_widget analogue_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN1LN"), SND_SOC_DAPM_INPUT("IN1LP"), SND_SOC_DAPM_INPUT("IN2LN"), -SND_SOC_DAPM_INPUT("IN2LP/VXRN"), +SND_SOC_DAPM_INPUT("IN2LP:VXRN"), SND_SOC_DAPM_INPUT("IN1RN"), SND_SOC_DAPM_INPUT("IN1RP"), SND_SOC_DAPM_INPUT("IN2RN"), -SND_SOC_DAPM_INPUT("IN2RP/VXRP"), +SND_SOC_DAPM_INPUT("IN2RP:VXRP"), SND_SOC_DAPM_MICBIAS("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0), SND_SOC_DAPM_MICBIAS("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0), @@ -537,14 +537,14 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "IN1R PGA", "IN1RP Switch", "IN1RP" }, { "IN1R PGA", "IN1RN Switch", "IN1RN" }, - { "IN2L PGA", "IN2LP Switch", "IN2LP/VXRN" }, + { "IN2L PGA", "IN2LP Switch", "IN2LP:VXRN" }, { "IN2L PGA", "IN2LN Switch", "IN2LN" }, - { "IN2R PGA", "IN2RP Switch", "IN2RP/VXRP" }, + { "IN2R PGA", "IN2RP Switch", "IN2RP:VXRP" }, { "IN2R PGA", "IN2RN Switch", "IN2RN" }, - { "Direct Voice", NULL, "IN2LP/VXRN" }, - { "Direct Voice", NULL, "IN2RP/VXRP" }, + { "Direct Voice", NULL, "IN2LP:VXRN" }, + { "Direct Voice", NULL, "IN2RP:VXRP" }, { "MIXINL", "IN1L Switch", "IN1L PGA" }, { "MIXINL", "IN2L Switch", "IN2L PGA" }, @@ -565,7 +565,7 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "Left Output Mixer", "Right Input Switch", "MIXINR" }, { "Left Output Mixer", "IN2RN Switch", "IN2RN" }, { "Left Output Mixer", "IN2LN Switch", "IN2LN" }, - { "Left Output Mixer", "IN2LP Switch", "IN2LP/VXRN" }, + { "Left Output Mixer", "IN2LP Switch", "IN2LP:VXRN" }, { "Left Output Mixer", "IN1L Switch", "IN1L PGA" }, { "Left Output Mixer", "IN1R Switch", "IN1R PGA" }, @@ -573,7 +573,7 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "Right Output Mixer", "Right Input Switch", "MIXINR" }, { "Right Output Mixer", "IN2LN Switch", "IN2LN" }, { "Right Output Mixer", "IN2RN Switch", "IN2RN" }, - { "Right Output Mixer", "IN2RP Switch", "IN2RP/VXRP" }, + { "Right Output Mixer", "IN2RP Switch", "IN2RP:VXRP" }, { "Right Output Mixer", "IN1L Switch", "IN1L PGA" }, { "Right Output Mixer", "IN1R Switch", "IN1R PGA" }, -- 1.6.3.3

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[alsa-devel] 48kHz SPDIF out on X-Fi?
by Lucas Marshall 04 Dec '09

04 Dec '09

Is it possible to get the optical SPDIF on an X-Fi Titanium to output at 48kHz instead of 96kHz? I've tried doing a rate convert in .asoundrc, adding a rate to the hw plug in there, etc. but no matter what it's still spewing 96kHz according to my test receiver. We would like to do a SPDIF-to-AES conversion, and our video/audio router won't take anything but 48kHz. :( Thanks in advance for any suggestions! -- Lucas Marshall Systems Administrator // Media Systems Lucasfilm 415-746-2861

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Re: [alsa-devel] [Alsa-user] m-audio delta 1010 inputs crackle at 96 kHz sample rate with all newer kernels
by francis keyes 03 Dec '09

03 Dec '09

The first kernel that broke was a newer 64 bit one. I switched to 32 bit Linux a few months later to try it out because I had other problems. However, this particular problem persists. On Wed, Dec 2, 2009 at 3:49 AM, Clemens Ladisch <clemens(a)alsa-project.org>wrote: > francis keyes wrote: > > The last kernel that worked properly for me was > > > > 2.6.24.7-1.rt21.1.fc9.ccrma.x86_64.rt > > > > (I am still using the same hardware but I no longer use 64 bit linux) > > So the first kernel that broke was the same version in 32 bits, or some > newer one? > > > Best regards, > Clemens >

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